similar to: Using hint priority with LDAP extensions and users

Hi, I am turning crazy. I try to integrate Asterisk 11.5.1 into Samba4 LDAP, but when I import the ldif file from contrib directory I get this error. ldbmodify -H /usr/local/samba/private/sam.ldb asterisk.ldif --option="dsdb:schema update allowed"=true ERR: (No such object) "objectclass: Cannot add cn=asterisk,cn=schema,cn=config, parent does not exist!" on DN

Hello list, I was going to buy this weekend a Wisip from http://www.pulverinnovations.com/, but jeff got out of stock and he wont have Wisip for the next 3 to 4 weeks. So I start searching for other wifi phones because I was really upset about it and I found IPC5000 from http://www.fahdtel.com/sip_phones.htm, I liked so much the pic that I email the guy and he send me the PDF with all the details

Hi, Is there a way to send and receive SMS over SIP protocol with Asterisk ? I mean, between two SIP phones like below... SIP_phone "A" (sending sms) ====> Asterisk ========>SIP_phone "B" (receiving sms) ... Is it possible ? If so, how could I do it ? Thanks, Angel. -------------- next part -------------- An HTML attachment was scrubbed... URL:

I see my own number(or remote called num) instead of caller id on incoming calls on my BT-102. but on Xlite everyything is OK. I'm using * latest CVS. - shabanip

Dear all i have one confusion about how to dial outgoing call through asterisk like when i press 0 i got dial ton of exchange for outgoing call my setup is [sip_phone]-----[*]----[mediant2k]-----[Avaya_PBX]------e1-----[Exchange_PSTN] now i want to setup whn i press 0 in my sip phone i got dialton of PSTN so i can call outside people is there any special configuration to give

Hello, I try configuring Quintum DX gateway as link to PSTN for *. Now, I can dial number which is connect to Quintum, and call is diverted to *. I don't know what I should set, if I want call from SIP_phone registred in Asterisk to PSTN via Quitnum. I set in sip.conf account for Quintum [sip_proxy-out] type=peer outboundproxy=QUINTUM_IP , and changed extensions.conf. When

Greetings. Attempting to connect Asterisk to LDAP database using res_config_ldap module. While trying to register sip client (Ekiga softphone), according to slapd.log, asterisk connects to LDAP server, asks for some attributes to modify (they do exist, and asterisk user has all permissions to do that, etc). And then asterisk application just crashes. Without ldap (using just static users'

I need to set up an office full of Cisco 7960 phones behind NAT with the server out in Colo. The first test phone registers fine, but the second one does not register. The first phone's registration looks like so: /SIP/Registry/3115552368 :64.169.xx.yyy:38836:3600:3115552368:sip:3115552368@64.169.xx.yyy:5060 When the second phone tries to register, it gets back a 404 not found. Not a

Hi, I just upgraded by 1.0.x home server to 1.2.2. Overall the upgrade went fine, but a strange problem has cropped up with the CALLERID name value of incoming calls from the X101P card. When an incoming call is presented (via Vonage ATA), the calledid value getting double quotes up from: -- Executing NoOp("Zap/1-1", """WIRELESS CALLE" <1404xxxxxxx>") in

Source: xen Version: Lack of copyright entry for VInay Sajip Severity: normal Dear Maintainer, I noticed that d/copyright lacks information for Vinay Sajip and his permissive license. You can see for which files it should be included: http://codesearch.debian.net/search?prev=0&q=the+name+of+Vinay+Sajip&skip=99, e.g. tools/python/logging/logging-0.4.9.2/* and

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I have build asterisk from latest CVS HEAD-05/09/05 with H323 support as described in README file. Open H.323 version v1.17.1 and PWLib v1.9.0 on Mandrake Linux 10.2 kernel-2.6.11 I tested it with following phones: -- XLite (SIP softphone) -- QMix SIP IP phone (PA168F) -- SJPhone (H323 softphone) -- QMix H323 IP phone (PA168F) -- FireFly (IAX2 softphone) Everything works fine except a problem

Hello I have problem with deliver dying with signal 11. I'm using postfix + dovecot devliver. If mailbox have many (100+) redirects in sieve or many other sieve rules deliver died. I have tested this in debian etch + dovecot 1.1.18 (compiled from sources) and debian lenny + dovecot 1.1.13 from backports. Sep 7 13:58:19 mail postfix/pipe[5964]: AC6835938: to=<test1 at mx.domain>,

I write the email again, the third time!!, cause the other two ones, I have had problems while sending them. I hope this time it works. Here is the email again: Hi (and sorry) everybody I'm starting with SIP and I wanted to ask some questions, perhaps silly ones, but I hope people can answer me! 1) Which codecs may I use? I want the SIP phones to call to the PSTN above all, but I have

Hi, I would like to know, if there's any R function, which allows calculation of probability values (0,1) from multivariate normal densities. I would be grateful for any output. Cheers, MG

I started SolidWorks (3D CAD program) with wine. As soon as I go from 2D sketch to 3D solid the screen is becoming black. One can see some lines as it is rotated. The same is for wireframe and rendered solids. 2D works very good. What can be the reason ? (SW is openGL based) Any help is welcome Marek Morzynski

Witam Sieve implementation in dovecot 1.2 have hard-coded limit for max redirect: SIEVE_DEFAULT_MAX_REDIRECTS 4. I have to rise this to much higher value, e.g. 64. Is there possibility to do this another way than recompiling source? Maybe adding an option to dovecot.conf in plugin section will be a good idea. -- Pozdrowienia Maciej Polewczynski Registered Linux user #117725 OGICOM Sp.

Dear sir I have setup Avaya with mediant with asterisk [sip_phone]---[ * ]---[mediant]---E1-trunk--[Avaya]---[analog_phone] This is my configuration when i call from SIP phone i got ringing sound of phone but whn i call from analog_phone behind avaya i didn't get ring sound of ring but SIP phone speaker ring why i am not able to hear ring sound from analog phone Regards

http://bugs.freedesktop.org/show_bug.cgi?id=13963 Summary: Menu buttons don't work on Malta's page. Product: swfdec Version: 0.5.5 Platform: Other URL: http://www.malta.poznan.pl/lodowisko/ OS/Version: All Status: NEW Severity: normal Priority: medium Component: plugin

I would like to load ApoAI.RData. During the operation of reading this data an error occurs. There is also a problem with STF file. > library (limma) > load("ApoAI.RData") Warning message: 'RG' looks like a pre-2.4.0 S4 object: please recreate it > objects() [1] "RG" > names(RG) [1] "R" "G" "Rb"